Telephony communications are increasingly being communicated across digital networks. As a result, it is becoming increasingly desirable to communicate voice over these networks too. Presently, this is accomplished by using Voice over Packet (VoP) systems that compress the voice in accordance with an International Telecommunication Union (ITU) standard. After being conveyed across the digital network, the voice packets are decompressed and used to reproduce a signal at the destination terminal which attempts to closely match the original signal. This solution significantly reduces the required bandwidth while maintaining high voice quality. For example, an uncompressed voice signal requires a bandwidth of 64,000 bits per second (bps). A compressed version of this same voice signal may be communicated with as little as 8,000 bps while still preserving the toll quality of the telephone call.
Each compression of an audible signal, which reduces the necessary bandwidth, also reduces the resolution of the signal and causes some distortion. A substantial bandwidth reduction below 8,000 bps may not be possible using prior art methods, without greatly affecting the quality of the voice signal.